Which grandstream phone should I buy, this is going to be for small
office for testing purposes.
I am on a budget, hoping to find someone here who has some used to
sell or point me in the direction of a seller.
Hi Mike,
If you're set on the Grandstreams, and it's just for testing the
Hi,
I am having some trouble with parked calls timing out.
In features.conf:
[general]
parkext = 800 ; What extension to dial to park
parkpos = 801-820 ; What extensions to park calls on
context=parkedcalls
parkingtime=120
After the Park timesout it calls the phone that the call was parked
On Wed, 12 Nov 2008, Mike wrote:
Hi All,
I have setup asterisk 1.4.22; so far everything good.
Except, I am still searching for voIP phones.
Which grandstream phone should I buy, this is going to be for small
office for testing purposes.
I am on a budget, hoping to find someone here who
Hello!
However I'm a newbie in Asterisk/VOIP/CM I would like to make sure that this
system design can work:
Cisco 2811 Voice Gateway - sip trunk1 - asterisk on linux box - síp trunk2 -
Cisco Call Manager 6.0
There is also a Siemens Hicom old pbx connected with QSIG to the Voice Gateway.
I
Daniel Johnson wrote:
[park-dial]
My park-dial is simply a 'goto' to a different part of my dial plan, in
this case, the part that handles incoming calls to the operator:
[park-dial]
exten = t,1,goto(office-hours,s,6)
== Auto fallthrough, channel 'SIP/551-b6d409d8' status is
I'll be rolling out a 1.6 installation within a couple of weeks that
will be a full featured PBX including MySQL realtime sip iax
users/peers, voicemail configs, queue_logs, cdrs, queues, and
queue_members. I'll post some data once I get it running...First I have
to get the old config (that
Hi all,
As OpenSIPS/OpenSER/SER are complementary SIP entities for many Asterisk
installations, maybe this topic will sound interesting to some of you.
Any contribution from the Asterisk users side (what they expect / how
they see OpenSIPS in a Asterisk env) may prove to be a valuable one.
For
Hi!
Is there somewhere a statement from Digium how long they will support
Asterisk 1.4?
thanks
klaus
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Hi!
What is the preferred way to make database lookups from within the dialplan?
I only know the MYSQL function from asterisk-addons. Are the other
methods too? (e.g. for postgresql, unixodbc)
thanks
klaus
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Firstly, I'm running Asterisk 1.4.4 on Solaris 10.
I have several different internal SIP phones all sharing a single IAX2
VoIP channel.
PHONES |- SIP/uLAW --| ASTERISK
|-- IAX2/g729 |VoIP/ISP
The g729 codec has been registered successfully and
Hi there!
I found solution about marking packets.
In sip.conf I put
...
tos_sip=cs5
tos_audio=ef
tos_video=af41
...
But when I test the packets with wireshark, the sip qos is set at 0x22 with
AF=41
Is the tos should be forever set at tos_sip=cs3?
Anyone could help me please?
Actually I would nat=yes always, even if clients are not behind NAT os
otherwise the clietn can put some garbage into the contact header (e.g.
IP address of an upstream provider) and influence routing.
The only thing were nat=yes is bad is if you have an asymmetric client.
I do not know any
On Thu, Nov 13, 2008 at 9:56 AM, Matt Gibson [EMAIL PROTECTED]wrote:
Which grandstream phone should I buy, this is going to be for small
office for testing purposes.
I am on a budget, hoping to find someone here who has some used to
sell or point me in the direction of a seller.
Hi
Klaus Darilion wrote:
Actually I would nat=yes always, even if clients are not behind NAT os
otherwise the clietn can put some garbage into the contact header (e.g.
IP address of an upstream provider) and influence routing.
No. There is a specific reason RFC 3261 says:
Registration
Alex Balashov wrote:
Klaus Darilion wrote:
Actually I would nat=yes always, even if clients are not behind NAT os
otherwise the clietn can put some garbage into the contact header (e.g.
IP address of an upstream provider) and influence routing.
No. There is a specific reason RFC 3261
On Thu, 2008-11-13 at 15:16 +0100, Klaus Darilion wrote:
What is the preferred way to make database lookups from within the dialplan?
The preferred method is to use func_odbc, which takes SQL queries and
builds custom dialplan functions from them. I've used it quite a bit,
and am very happy
Hi,
you yould also use DBQuery (does only support mysql) - take a look at
http://www.voip-info.org/wiki/view/Asterisk+cmd+DBQuery (it does also
contain a cdr backend to write customzied cdr entries to the database)
regards,
Wolfgang
Klaus Darilion schrieb:
Hi!
What is the preferred way to
I have a 9 agents implementation with a queue, conference, transfers, some
problems at the beginning with no hangup detected in some calls,
reinstallation of dahdi and everithing is ok now.
2008/11/13 Sherwood McGowan [EMAIL PROTECTED]
I'll be rolling out a 1.6 installation within a couple of
On Thu, Nov 13, 2008 at 10:19 AM, Alex Balashov
[EMAIL PROTECTED]wrote:
Alex Balashov wrote:
Klaus Darilion wrote:
Actually I would nat=yes always, even if clients are not behind NAT os
otherwise the clietn can put some garbage into the contact header (e.g.
IP address of an upstream
Steve Totaro wrote:
Alex is going to cling to to the RFC as if it were the gospel, and not
look at what would essentially be a good thing.
The RFC is not the gospel, but nor is it just a request for comment,
historical nomenclature aside.
It is the de facto standard for the implementation
On Thu, Nov 13, 2008 at 12:19 PM, Alex Balashov
[EMAIL PROTECTED]wrote:
Steve Totaro wrote:
Alex is going to cling to to the RFC as if it were the gospel, and not
look at what would essentially be a good thing.
The RFC is not the gospel, but nor is it just a request for comment,
You're right, all that verbose book-learnin' and complex protocol
implementations definitely don't belong together.
Steve Totaro wrote:
What is Asterisk designed to be? A PBX. (yes that is a period)
That question will fetch many answers depending on who you're talking
to. It is used for a
Alex Balashov schrieb:
Klaus Darilion wrote:
Actually I would nat=yes always, even if clients are not behind NAT os
otherwise the clietn can put some garbage into the contact header (e.g.
IP address of an upstream provider) and influence routing.
No. There is a specific reason RFC
Alex Balashov schrieb:
Steve Totaro wrote:
Alex is going to cling to to the RFC as if it were the gospel, and not
look at what would essentially be a good thing.
The RFC is not the gospel, but nor is it just a request for comment,
historical nomenclature aside.
It is the de facto
Klaus Darilion wrote:
Of course we know that we should implement RFC conform. But RFC 3261 has
ignored the fact that the Internet is full of NATs and standard conform
implementations can not work. This in the case of SIP it necessary to
break the RFC.
By default?
NAT itself is a hack;
Klaus Darilion wrote:
This is a different scenario. In this case of course I want the public
IP of the client, not of the load balancer. So, yes - in this case
nat=no is useful for Asterisk. Nevertheless I ignore the IP provided by
the client in the contact header completely - I always use
On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote:
Is there somewhere a statement from Digium how long they will support
Asterisk 1.4?
There is no statement, because we haven't even discussed when the EOL for
1.4 will be reached. Certainly that means it won't happen for at least the
Wolfgang Pichler schrieb:
Hi,
you yould also use DBQuery (does only support mysql) - take a look at
http://www.voip-info.org/wiki/view/Asterisk+cmd+DBQuery (it does also
contain a cdr backend to write customzied cdr entries to the database)
hi wolfgang!
Have you programmed this
Hi Jared!
Thanks for the info - looks very flexible - you only have to edit 4
configuration files for a simple query :-)
just a few questions:
The ODBC library is unixodbc?
How does it compare to the other solutions in terms of performance? e.g.
(I have to make several queries for each call
Alex Balashov schrieb:
Klaus Darilion wrote:
This is a different scenario. In this case of course I want the public
IP of the client, not of the load balancer. So, yes - in this case
nat=no is useful for Asterisk. Nevertheless I ignore the IP provided by
the client in the contact
Alex Balashov schrieb:
Klaus Darilion wrote:
Of course we know that we should implement RFC conform. But RFC 3261 has
ignored the fact that the Internet is full of NATs and standard conform
implementations can not work. This in the case of SIP it necessary to
break the RFC.
By
Hi,
yes - i did have done it myself.
The main difference is that the dbquery application does use the
res_mysqlpool - which is a small piece of code which does handle a pool
of connections to one or more mysql servers. So you can make a fault
tolerant system by using two master - master
Mark Michelson wrote:
c james wrote:
Mark Michelson wrote:
c james wrote:
Mark Michelson wrote:
c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 # ./configure
Klaus Darilion wrote:
Very often the authentication between gateway and proxy is done based on
IP address.
Now, the customers SIP client at 3.3.3.3 REGISTERs to the proxy with
Contact: sip:[EMAIL PROTECTED]
If now there is an incoming call to the customer, the Proxy/Asterisk
will
I'm doing a new install for an old customer. The customer is running a
custom version of Asterisk based on version 1.2.7.1. It works for them --
aside from a memory leak requiring a restart once every couple of
months...
I think the corresponding version of Zaptel is 1.2.5, but I'd like to
On Thu, 13 Nov 2008, Steve Edwards wrote:
I'm doing a new install for an old customer. The customer is running a
custom version of Asterisk based on version 1.2.7.1. It works for them --
aside from a memory leak requiring a restart once every couple of
months...
I think the corresponding
c james wrote:
Mark Michelson wrote:
c james wrote:
Mark Michelson wrote:
c james wrote:
Mark Michelson wrote:
c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 #
Hello all,
I have the following working (somewhat) setup:
TELCO
|
|
E1 (30 Chan -- TE210 SPAN 2)
|
|
Asterisk box 1.6 with
DAHDI drivers loaded
Digium TE210p
|
|
E1 (30 Chan -- TE210 SPAN 1)
|
|
NEC PBX
From the NEC system I can make calls out. From a line
In theory ParkAndAnnounce has a lot of usefulness, however, that we've had
very little success with application...Our application is similiar to
the local Walgreens pharmacy.. Dr. Calls in, selects the Im a doctor with
a prescription option...call is parked, and announcement overhead is
hi;
I am attempting to find a way to install pbxinaflash into hypervm, so
to find a templet for hypervm with pbxinaflash. Do you guys know of
any i can get or can someone design one for me so i can install that
templet in a hypervm virtual machine? I can't appear to figure out
where i
Hi there,
Is there anyone that has already use asterisk 1.4 with openLDAP?
Could you give me the link where you obtain the res_config_ldap.c and the
asterisk schema please? a
regards,
Jude
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